基于谱减和LMS的自适应语音增强

针对宽带噪声背景下的语音增强问题,将短时语音视为非平稳或宽平稳信号,基于谱减法和自适应滤波的最小均方（LMS）算法,提出了一种FIR型自适应滤波算法（SSLMS）：用减谱法由短时噪声观测语音估计期望信号,作为滤波器输出信号的参考信号;用滤波器的输出与参考信号的差值为误差信号,用LMS算法求得滤波器权系数修正量,并修正滤波器。权系数最速下降调整中,采用了归一化LMS、符号LMS、块LMS技术,以简化保证权系数收敛的步长选择、减少权系数修正的运算量,从而提高自适应速度。对不同的语音在各种信噪比下仿真实验,并与改进的谱减法比较,结果表明,该法增强效果优于谱减法;在信噪比为3 dB时该法的增强效果仍然令人满意。     

A new adaptive center weighted median filter for suppressing impulsive noise in images

In this work, a new adaptive center weighted median (ACWM) filter is proposed for improving the performance of median-based filters, preserving image details while effectively suppressing impulsive noise. The proposed filter is an adaptive CWM filter with an adjustable central weight obtained by partitioning the observation vector space. To obtain the optimal weight for each block, the efficient scalar quantization (SQ) method is used to partition the observation vector space. The center weight within each block is obtained by using a learning approach based on the least mean square (LMS) algorithm. The noise filtering procedure is progressively applied through several iterations so that the mean square error of the output can be minimized. Experimental results have demonstrated that the proposed filter outperforms many well-accepted median-based filters in terms of both noise suppression and detail preservation. The proposed new filter also provides excellent robustness at various percentages of impulsive noise.     

Frequency-response-shaped LMS adaptive filter

A new LMS algorithm is introduced for improved performance when a sinusoidal input signal is corrupted by correlated noise. The algorithm is based on shaping the frequency response of the transversal filter. This shaping is performed on-line by the inclusion of an additional term similar to the leakage factor in the adaptation equation of leaky LMS. This new term, which involves the multiplication of the filter coefficient vector by a matrix, is calculated in an efficient manner using the FFT. The proposed adaptive filter is shown analytically to converge in the mean and mean-square sense. The filter is also analyzed in the steady state in order to show the frequency-response-shaping capability. Simulation results illustrate that the performance of the frequency-response-shaped LMS (FRS-LMS) algorithm is very effective even for highly correlated noise.     

免疫算法自适应滤波器对白噪声的抑制

实际信号经常会受到白噪声及高次谐波的影响,由于白噪声频谱分布在整个实数域,常用的滤波器很难将其滤除。讲述了自适应滤波器的原理及用免疫算法自适应滤波器,对白噪声及高次谐波进行抑制的方法。通过免疫算法对自适应滤波器的权向量进行优化,并用均值滤波的方法对自适应滤波器的滤波结果进行进一步滤波,然后用MATLAB对该算法进行仿真。将免疫算法自适应滤波器的仿真结果和LMS滤波算法的仿真结果进行比较,表明免疫算法自适应滤波器能对白噪声及高次谐波进行有效的抑制。     

基于并行技术和流水线的LMS自适应滤波算法

针对现有自适应滤波算法中数据处理效率低的问 题,提出了基于并行技术和流水线的最小均方误差（Least mean square,LMS）自适应滤波算法。该算法构建基 于并行技术的多输入多输出滤波器结构,成倍提高系统滤波处理速度;设计基于流水线的LMS 自适应滤波权系数求解方法,有效改善了权系数计算效率。最后利用现场可编程门阵列(Field programmable gate array,FPGA)对该算法进行了验 证,结果表明,对于四级并行流水线四阶LMS自适应滤波器,其数据处理速率提高了约8倍,在相同的数据处理速率下,其功耗可降低约84%,从而提高了LMS自适应滤波处理速率,降低了系统功耗,实现了高速、超高速数据流的实时自适应滤波 处理。     

On the convergence of LMS filters under periodic signals

The response of the Least Mean Square (LMS) algorithm to deterministic periodic inputs is considered. Under these conditions, initial values of the tap-weight vector can be identified that lead to periodic responses of LMS filters. The stability of these periodic responses determines the long-term convergence of the filter. This analysis presents some advantages over the classical studies based on the correlation matrix, because it leads to more accurate results and a better understanding of the filter operation. It is also shown that such an operation does not change essentially for more realistic inputs, as when the desired response is perturbed with a zero-mean random signal. Finally, to validate the obtained results, some simulations and experiments have been conducted for an adaptive noise canceller.     

An adaptive unscented Kalman filter approach to secure state estimation for wireless sensor networks

Wireless sensor networks are vulnerable to false data injection attacks, which may mislead the state estimation. To solve this problem, this paper presents a chi-square test-based adaptive secure state estimation (CTASSE) algorithm for state estimation and attack detection. Taking advantage of Kalman filters, attack signal together with process noise or measurement noise are described as total white Gaussian noise with uncertain covariance matrix. The chi-square test method is used in the adaptation of the total noise covariance and attack detection. Then, a standard adaptive unscented Kalman filter (UKF) is used for the state estimation. Finally, simulation results show that the proposed CTASSE algorithm performs better than other UKFs in state estimation and is also effective in real-time attack detection.     

滤波器组自适应语音增强方法

双麦克风噪声抵消应用中,由于交叉串的存在,传统自适应算法降噪性能受到很大的影响。为了提高双麦克风算法降噪性能,使用两级自适应滤波系统消除交叉串扰问题。为提高自适应滤波器收敛性能,采用主从结构LMS算法自适应调节步长因子。同时为了适合窄带处理算法,将输入信号进行子带分析预处理,对每个子带独立进行抗交叉串绕自适应处理,将各子带增强信号合并得到增强语音信号。实验结果表明,该方消噪量大,语音损伤小,语音增强效果显著。     

Partition-based fuzzy median filter based on adaptive resonance theory

This paper presents a novel partition-based fuzzy median filter for noise removal from corrupted digital images. The proposed filter is obtained as the weighted sum of the current pixel value and the output of the median filter, where the weight is set by using fuzzy rules concerning the state of the input signal sequence to indicate to what extent the pixel is considered to be noise. Based on the adaptive resonance theory, the authors developed a neural network model and created a new weight function where the neural network model is employed to partition the observation vector. In this framework, each observation vector is mapped to one of the M blocks that form the observation vector space. The least mean square (LMS) algorithm is applied to obtain the optimal weight for each block. Experiment results have confirmed the high performance of the proposed filter in efficiently removing impulsive noise and Gaussian noise.     

Accuracy evaluation of fixed-point based LMS algorithm

The implementation of adaptive filters with fixed-point arithmetic requires computation quality evaluation. The accuracy may be determined by computing the global quantization noise power at the system output. In this paper, a new model for evaluating analytically the global noise power in LMS-based algorithms is presented. Thus, the model is developed for LMS and NLMS algorithms. The accuracy of our model is analyzed by simulations.     

基于信息补偿的自适应联邦滤波算法

针对采用标准卡尔曼滤波器必须知道系统噪声统计特性的局限性,研究了一类系统噪声未知情况下的自适应联邦滤波方法,指出了自适应滤波方法应用于联邦结构时应当注意的问题,提出了一种基于信息补偿的自适应联邦滤波算法。SINS／BDS／GPS组合导航系统的仿真结果表明,该方法可以有效抑制系统噪声未知情况下的滤波发散现象,提高了滤波的稳定性和估计性能。     

基于LabVIEW的自适应滤波器设计与研究

针对自适应滤波器编程复杂,难以按照虚拟仪器系统的形式来测试工程应用中的实际性能等问题。文中利用LabVIEW8.6提供的自适应滤波器工具包,设计了基于最小均方误差算法、递推最小二乘算法的自适应滤波器,并对影响两种算法的参数对滤波器的敏感性进行了分析;进而,利用音频信号验证了滤波器性能。仿真结果表明,所设计的自适应滤波器功能全面,人机交互界面良好,便于工程技术人员快速开发,具有较好的工程实用价值。     

考虑输出饱和约束的自适应FIR滤波器设计

针对Bershad算法中滤波器输出信号畸变的问题,提出了一种新的考虑输出饱和约束的自适应有限脉冲响应(Finite-impulse-response, FIR)滤波器设计方法,并给出了它的最小均方(Least mean square, LMS)算法实现.理论证明和数值仿真验证了本文算法的有效性.与Bershad算法相比,按照本文算法设计的自适应FIR滤波器不仅严格满足饱和约束条件,输出信号也光滑无畸变,从而克服了Bershad算法的滤波器输出畸变问题.     

基于LMS算法自适应噪声抵消器的分析研究

自适应信号处理的理论和技术已经成为人们常用的语音去噪技术,而Matlab为其提供了更为方便快捷的方法来对语音信号进行消噪处理。通过介绍自适应滤波器原理,在对自适应滤波器相关理论研究的基础上,重点研究了LMS自适应滤波算法,并对LMS自适应算法进行了分析,用Matlab对其进行了仿真和实现。     

基于直接学习法的一种新预失真器

为了克服宽带信号经过记忆放大器的非线性失真,针对有记忆非线性功放的多项式模型,提出了一种新的基于直接学习法的自适应算法.该算法采用无记忆预失真器的级联扩展,具有横向滤波器结构,与记忆多项式有相似的线性化效果.并且针对信号噪声对自适应算法的扰动和收敛速度慢等缺点,采用归一化LMS算法加以改进.在非线性功放的记忆多项式模型下,通过宽带信号验证了基于直接学习法的记忆型预失真器算法的有效性.     

隐Markov模型的多算法交叉耦合自适应算法

以离子通道信号重构为例，扩展HMM为矢量隐Markov模型，利用随机逼近原理对期望最大算法进行自适应改造，估计离子通道的动力学特征参数；递归辅助变量算法估计背景噪声的统计特征；卡尔曼滤波预测背景噪声；三种算法交叉耦合重构离子通道信号。该算法能够克服滤波器和背景噪声的影响，在低信噪比情况下得到了较高精度的估计参数和重构信号，具有鲁棒性和一致收敛特性。     

基于CRV-LMS算法的语音增强技术的研究

在语音增强系统中,传统的LMS算法存在剩余均方误差随有用信号功率线性增加的问题。本文提出CRV-LMS算法,该算法首先设计组合滤波器,然后在滤波器中采用变步长系数,通过设计变步长更新律,从而使新权值更新收敛系数与输出功率成反比,达到提高语音增强系统中性能的目的。     

多尺度小波分解下的自适应语音消噪算法研究

将小波变换的理论引入到自适应语音消噪系统中,分析了多尺度小波分解下的LMS自适应消噪算法（MSWD-LMS）的原理,该算法将输入向量分解到多尺度空间,减小了自适应滤波器输入向量自相关矩阵的谱动态范围;将变步长LMS算法与多尺度小波变换的思想结合,提出了一种新的小波自适应语音消噪算法（MSWD-VSS-LMS）,新算法既减少了输入向量自相关矩阵条件数,又克服了固定步长LMS算法在收敛速度与收敛精度方面与步长因子μ的矛盾,获得了更好的语音信号处理的收敛速度和稳定性。仿真结果表明新算法取得了较好的效果。     

Adaptive Gaussian filters for nonlinear state estimation with one-step randomly delayed measurements

This paper proposes new algorithms of adaptive Gaussian filters for nonlinear state estimation with maximum one-step randomly delayed measurements. The unknown random delay is modeled as a Bernoulli random variable with the latency probability known a priori. However, a contingent situation has been considered in this work when the measurement noise statistics remain partially unknown. Due to unavailability of the complete knowledge of measurement noise statistics, the unknown measurement noise covariance matrix is estimated along with states following: (i) variational Bayesian approach, (ii) maximum likelihood estimation. The adaptation algorithms are mathematically derived following both of the above approaches. Subsequently, a general framework for adaptive Gaussian filter is presented with which variants of adaptive nonlinear filters can be formulated using different rules of numerical approximation for Gaussian integrals. This paper presents a few of such filters, viz., adaptive cubature Kalman filter, adaptive cubature quadrature Kalman filter with their higher degree variants, adaptive unscented Kalman filter, and adaptive Gauss–Hermite filter, and demonstrates the comparative performance analysis with the help of a nontrivial Bearing only tracking problem in simulation. Additionally, the paper carries out relative performance comparison between maximum likelihood estimation and variational Bayesian approaches for adaptation using Monte Carlo simulation. The proposed algorithms are also validated with the help of an off-line harmonics estimation problem with real data.     

激光陀螺信号解调中自适应滤波器的设计

在激光陀螺信号解调领域中,在满足高精度的前提下如何降低滤波器的延迟一直是相关院所的研究重点。针对此问题,研究了一种新的激光陀螺滤波处理的方法。这种方法采用LMS自适应滤波器原理,分别把机械抖动抖反馈信号作为滤波器的基本输入,把机抖信号、随机噪声和白噪声作为滤波器的参考信号,然后通过FPGA进行数字滤波以及外围控制,最后给出了滤波器的算法实现以及硬件框图。实验结果表明,LMS自适应滤波器有很好的解调效果,经过滤波后的计数值差值在±1个数以内,且延时为1 ms。